[Asterisk-Dev] changing codec during call

[Daniel Bichara]

Hi all,

Regarding changing codec during call:

In a IAX connection, could I change the codec from GSM to G.711 during 
the call if I detect a fax?

Thanks in advance

Daniel

Steve Kann wrote:

> Steve Underwood wrote:
>
>> Jesse Kaijen wrote:
>>
>>> The reason I like to use the IAX-protocol is that the new 
>>> jitterbuffer is
>>> based on the E-MOS algorithm (PLEASE CORRECT ME IF I'M WRONG). 
>>> **see_below**
>>>  
>>>
>> Why is that good? the new jitter buffer is intended to be an 
>> improvement, but it is certainly not state of the art. Far better 
>> results are possible.
>
>
>
> Are you talking about the JB, our the WSOLA timescale modification 
> stuff you talked about before?
>
> I think that, even without changing the architecture of things a lot, 
> we could do the WSOLA stuff by adding a "timescale" parameter to 
> frames, and during translation or wherever we would do PLC initially, 
> we could adjust the playout based on that timescale.  The jitterbuffer 
> could pretty easily add the data to drive this (where, it would do 
> something like adjust 10 frames up or down 10% whenever it's dropping 
> or interpolating now).
>
> -SteveK
>
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